Control method of adaptive array and adaptive array apparatus

Data processing: measuring – calibrating – or testing – Measurement system – Measured signal processing

Reexamination Certificate

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C381S092000

Reexamination Certificate

active

06449586

ABSTRACT:

BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a control method of adaptive array and an adaptive array apparatus for receiving a signal in spatially selective manner employing a plurality of sensors.
2. Description of the Related Art
In a field of obtaining a voice signal, sonar, radio communication and so forth, in order to receive only a specific signal among a plurality of signal sources, a voice enhancing device employing an adaptive microphone array, a radio signal transmitting and receiving device employing an adaptive antenna array and so forth are known as application of an adaptive array technology.
As sensor, a microphone, an ultrasonic sensor, a sonar receiver, antenna and so forth may be employed. Discussion will be given hereinafter for the case where the microphone is used as the sensor.
Here, for simplification of disclosure, consideration is given for the case where the microphones are aligned with equal interval. On the other hand, a target sound source is considered to be located sufficiently distant from a line, on which the microphones are arranged. Also, a situation where the target sound source is arranged perpendicularly to the line on which the microphones are aligned.
A microphone array forms a spatial filter by summing signals sounded by a plurality of microphones after filtering. By such spatial filter, an environmental noise can be suppressed to permit reception of signal arriving from a predetermined direction, namely only a target sound. The adaptive microphone array is a microphone array which adaptively varies spatial filtering characteristics. As constructions of the adaptive microphone array, a construction disclosed in “Generalized Side Lobe Canceller”, IEEE, Transactions on Antennas and Propagation, Vol. 30, No. 1, 1982, pp 27 to 34 (hereinafter referred to as “publication 1”), a construction disclosed in IEEE, Transactions on Antennas and Propagation, Vol. 40, No. 9, 1992, pp 1093 to 1096 (hereinafter referred to as “publication 2”), a construction disclosed in Paper of The Institute of Electronics. Information and Communication Engineers, Vol. 79, No. 9, 1966, pp 1516 to 1524 (hereinafter referred to as “publication 3”), a construction shown in “Frost Beam Former”, IEEE, Processing of IEEE, Vol. 60, No. 8, 1972, pp 926 to 935, a construction disclosed in IEEE, Processings of International Conference on Acoustics, Speech and Signal Processing 94, 1994, pp IV-267 to 272 (hereinafter referred to as “publication 5”) and so forth are known.
Here, discussion will be given for operation of the construction of the publication
3
as typical construction, with reference to the drawing.
FIG. 35
shows a signal processing portion of an adaptive array of the publication 3, when M microphones are employed. Signals of a microphone group
1
m
(m=0, 1, . . . , M−1) are converted from analog signals into digital signals, respectively. This digital signal group (hereinafter referred to as microphone signal group) is subject to signal processing to extract a target signal.
The conventional adaptive array device is constructed with a fixed beam former
2
, a blocking matrix
20
, a multi-input canceller
30
. Hereinafter, each of the fixed beam former
2
, the blocking matrix
20
and the multi-input canceller
30
will be discussed individually.
As the fixed beam former, a delay and sum beam former which delays and sums signals received from the microphone group and a filter and sum beam former which filters and sums the signals received from the microphone group. Such fixed beam former has been disclosed in D. H. Johnson and D. E. Dudgeon, “Array Signal Processing” (Prentice Hall, Englewood Cliffs, 1993, Chapter 4 (hereinafter referred to as “publication 6”). Here, the operation will be discussed in terms of the delay and sum beam. The delay and sum beam former can be expressed by the following expression (1).
g



(
k
)
=

m
=
0
M
-
1



f
m



x
m



(
k
-
r
m
)
(
1
)
wherein k is a sample number in a time axis, and rm is a delayed sample number of respective microphone signals xm(k). g(k) is an output signal of the fixed beam former
2
, xm(k) is an output signal of a microphone Im, fm is a coefficients corresponding to the microphone signal in the fixed beam former.
The delay and sum beam former calculates and outputs a sum of the signals multiplied with the coefficients fm with delaying respective microphone signals xm(k) for rm samples. Each delay period rm is set to synchronize the phase of the target signal in a signal xm (k−rm) which is generated by delaying output signals of respective microphones Im. As a result, upon summing xm (k−rm) (m=0, 1, . . . , M−1), the target signal is enhanced. On the other hand, interference signals arrive from directions other than target signal. In the signal xm (k−rm) which is generated by delaying the output signal of each microphone, a phase is significantly different. Upon summing, the signals are canceled with each other to attenuate. Accordingly, in the output of the fixed beam former, the target signal is enhanced and the interference signal is attenuated.
Next, the blocking matrix
20
will be discussed with reference to FIG.
35
.
The blocking matrix
20
is constructed with a fixed beam former
3
, a delay group
4
m
(m=0, 1, M−1), an adaptive filter group
5
m
(m=0, 1, . . . , M−1), a subtractor group
6
m
(m=0, 1, . . . , M−1). The blocking matrix
20
is employed for adaptive signal processing to transmit a signal group, in which the target signal is attenuated and the signals other than the target signal are enhanced to the multi-input canceller
30
.
As a filter structure of the adaptive filter group
5
m
of the blocking matrix
20
, a finite impulse response (FIR) filter, an infinite impulse response (IIR) filter, lattice filter and so forth can be employed. Here, discussion will be given for the case where the FIR filter is employed.
The fixed beam former
3
receives the signal group from the microphone
1
and outputs a signal, in which the target signal is enhanced and the interference signals are attenuated by signal processing similar to the fixed beam former
2
. The output signal of the fixed beam former
3
becomes an input signal common to the adaptive filters
5
m
. Each delay
4
m
receives the output signal of corresponding one of the microphone
1
m
to transmit the delayed signal to corresponding one of the subtractor
6
m
. Each of the adaptive filters
5
m
receives the output signal of the fixed beam former
3
and transmits an output signal, in which a tap coefficients contained in the corresponding adaptive filter
5
m
is convoluted, to the corresponding subtractor
6
m
. Each subtractor
6
m
subtracts the output signal of the corresponding adaptive filter
5
m
from the output signal of the corresponding delay
4
m
. A result of subtraction of the subtractor
6
m
is transmitted to the corresponding adaptive filter
7
m
of the multi-input canceller
30
as the output signal of the blocking matrix
20
, and, in conjunction therewith, to the corresponding adaptive filter
5
m
for updating the tap coefficients.
A delay period of the delay
4
m
is set so that phases of a target signal component in the output of the delay 4 m and a target signal component in the output of the adaptive filter
5
m
are consistent with each other. For example, the delay period of the delay
4
m
may be set at a period as a sum of a group delay period of the fixed beam former
3
and a period of about one fourth to one half of a period corresponding to a tap number of the adaptive filter
5
m.
The process in the blocking matrix can be expressed by the following expression (2)
y
m
(k)=x
m
(k−P)−H
m
T
(k)D(k)(m=0,1, . . . , M−1)  (2)
wherein ym(k) is an output signal of the subtractor group
6
m
, xm(k−P) is an output signal of the delay
4
m
, P is a delay period of

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